56d945233d
We now have two downstream users of stun.h, so it appears to be generally usable. I put this in a new dir networking/, but I'm open to suggestions here (maybe some things in api/ should move in there). I checked what our downstream users are actually using, and it's cricket::ComputeStunCredentialHash cricket::<constants> cricket::TurnMessage cricket::GetStunErrorResponseType cricket::StunAttribute::CreateAddress cricket::StunErrorCodeAttribute cricket::StunByteStringAttribute StunAttribute::CreateUnknownAttributes cricket::TurnErrorType cricket::StunMessage I reckoned that was pretty much everything in stun.h, so I didn't bother splitting it up. They don't use every function and constant in there, but all _types_ of functions and constants, so for the sake of coherence I don't think it makes sense to split it. There's some old stuff in there like GTURN which could arguably be split out, but it should likely go away soon anyway, so I don't think it's worth the effort. Steps: 1) land this 2) update downstream to point to the new header and target 3) remove p2p/base:stun_types. Bug: webrtc:11091 Change-Id: I1f05bf06055475d25601197ec6fefb8d3b55e8e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159923 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29822}
636 lines
19 KiB
Plaintext
636 lines
19 KiB
Plaintext
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("//build/config/linux/pkg_config.gni")
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import("../webrtc.gni")
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group("media") {
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deps = []
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if (!build_with_mozilla) {
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deps += [
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":rtc_media",
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":rtc_media_base",
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]
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}
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}
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config("rtc_media_defines_config") {
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defines = [ "HAVE_WEBRTC_VIDEO" ]
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}
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rtc_library("rtc_h264_profile_id") {
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visibility = [ "*" ]
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sources = [
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"base/h264_profile_level_id.cc",
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"base/h264_profile_level_id.h",
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]
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deps = [
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"..:webrtc_common",
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"../rtc_base",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_source_set("rtc_media_config") {
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visibility = [ "*" ]
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sources = [
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"base/media_config.h",
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]
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}
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rtc_library("rtc_vp9_profile") {
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visibility = [ "*" ]
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sources = [
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"base/vp9_profile.cc",
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"base/vp9_profile.h",
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]
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deps = [
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"..:webrtc_common",
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"../api/video_codecs:video_codecs_api",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/system:rtc_export",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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rtc_library("rtc_media_base") {
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visibility = [ "*" ]
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defines = []
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libs = []
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deps = [
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":rtc_h264_profile_id",
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":rtc_media_config",
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":rtc_vp9_profile",
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"..:webrtc_common",
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"../api:array_view",
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"../api:audio_options_api",
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"../api:media_stream_interface",
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"../api:rtc_error",
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"../api:rtp_parameters",
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"../api:scoped_refptr",
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"../api/audio_codecs:audio_codecs_api",
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"../api/crypto:frame_decryptor_interface",
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"../api/crypto:frame_encryptor_interface",
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"../api/crypto:options",
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"../api/transport:stun_types",
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"../api/transport/media:media_transport_interface",
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"../api/transport/rtp:rtp_source",
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"../api/video:video_bitrate_allocation",
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"../api/video:video_bitrate_allocator_factory",
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"../api/video:video_frame",
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"../api/video:video_frame_i420",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../common_video",
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"../modules/audio_processing:audio_processing_statistics",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../p2p:stun_types",
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"../rtc_base",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:sanitizer",
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"../rtc_base:stringutils",
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"../rtc_base/synchronization:sequence_checker",
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"../rtc_base/system:file_wrapper",
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"../rtc_base/system:rtc_export",
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"../rtc_base/third_party/sigslot",
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"../system_wrappers:field_trial",
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"//third_party/abseil-cpp/absl/algorithm:container",
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"//third_party/abseil-cpp/absl/strings",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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sources = [
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"base/adapted_video_track_source.cc",
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"base/adapted_video_track_source.h",
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"base/audio_source.h",
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"base/codec.cc",
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"base/codec.h",
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"base/delayable.h",
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"base/media_channel.cc",
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"base/media_channel.h",
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"base/media_constants.cc",
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"base/media_constants.h",
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"base/media_engine.cc",
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"base/media_engine.h",
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"base/rid_description.cc",
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"base/rid_description.h",
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"base/rtp_data_engine.cc",
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"base/rtp_data_engine.h",
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"base/rtp_utils.cc",
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"base/rtp_utils.h",
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"base/stream_params.cc",
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"base/stream_params.h",
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"base/turn_utils.cc",
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"base/turn_utils.h",
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"base/video_adapter.cc",
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"base/video_adapter.h",
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"base/video_broadcaster.cc",
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"base/video_broadcaster.h",
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"base/video_common.cc",
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"base/video_common.h",
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"base/video_source_base.cc",
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"base/video_source_base.h",
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]
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}
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rtc_library("rtc_constants") {
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defines = []
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libs = []
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deps = []
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sources = [
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"engine/constants.cc",
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"engine/constants.h",
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]
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}
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rtc_library("rtc_simulcast_encoder_adapter") {
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visibility = [ "*" ]
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defines = []
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libs = []
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sources = [
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"engine/simulcast_encoder_adapter.cc",
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"engine/simulcast_encoder_adapter.h",
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]
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deps = [
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"../api:fec_controller_api",
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"../api:scoped_refptr",
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"../api/video:video_codec_constants",
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"../api/video:video_frame",
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"../api/video:video_frame_i420",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:rtc_software_fallback_wrappers",
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"../api/video_codecs:video_codecs_api",
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"../modules/video_coding:video_codec_interface",
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"../modules/video_coding:video_coding_utility",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/experiments:rate_control_settings",
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"../rtc_base/synchronization:sequence_checker",
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"../rtc_base/system:rtc_export",
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"../system_wrappers",
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"../system_wrappers:field_trial",
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"//third_party/abseil-cpp/absl/types:optional",
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"//third_party/libyuv",
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]
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}
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rtc_library("rtc_encoder_simulcast_proxy") {
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visibility = [ "*" ]
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defines = []
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libs = []
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sources = [
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"engine/encoder_simulcast_proxy.cc",
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"engine/encoder_simulcast_proxy.h",
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]
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deps = [
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":rtc_simulcast_encoder_adapter",
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"../api/video:video_bitrate_allocation",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:video_codecs_api",
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"../modules/video_coding:video_codec_interface",
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"../rtc_base/system:rtc_export",
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]
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}
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rtc_library("rtc_internal_video_codecs") {
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visibility = [ "*" ]
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allow_poison = [ "software_video_codecs" ]
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defines = []
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libs = []
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deps = [
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":rtc_constants",
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":rtc_encoder_simulcast_proxy",
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":rtc_h264_profile_id",
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":rtc_media_base",
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":rtc_simulcast_encoder_adapter",
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"../:webrtc_common",
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"../api/video:encoded_image",
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"../api/video:video_bitrate_allocation",
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"../api/video:video_frame",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:rtc_software_fallback_wrappers",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../call:video_stream_api",
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"../modules:module_api",
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"../modules/video_coding:video_codec_interface",
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"../modules/video_coding:webrtc_h264",
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"../modules/video_coding:webrtc_multiplex",
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"../modules/video_coding:webrtc_vp8",
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"../modules/video_coding:webrtc_vp9",
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"../rtc_base:checks",
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"../rtc_base:deprecation",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/system:rtc_export",
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"../test:fake_video_codecs",
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"//third_party/abseil-cpp/absl/strings",
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]
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sources = [
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"engine/fake_video_codec_factory.cc",
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"engine/fake_video_codec_factory.h",
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"engine/internal_decoder_factory.cc",
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"engine/internal_decoder_factory.h",
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"engine/internal_encoder_factory.cc",
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"engine/internal_encoder_factory.h",
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"engine/multiplex_codec_factory.cc",
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"engine/multiplex_codec_factory.h",
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# TODO(bugs.webrtc.org/7925): stop exporting this header once downstream
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# targets depend on :rtc_encoder_simulcast_proxy directly.
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"engine/encoder_simulcast_proxy.h",
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]
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}
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rtc_library("rtc_audio_video") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
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defines = []
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libs = []
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deps = [
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":rtc_constants",
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":rtc_media_base",
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"..:webrtc_common",
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"../api:call_api",
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"../api:libjingle_peerconnection_api",
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"../api:rtp_parameters",
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"../api:scoped_refptr",
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"../api:transport_api",
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"../api/audio:audio_mixer_api",
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"../api/audio_codecs:audio_codecs_api",
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"../api/task_queue",
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"../api/transport:bitrate_settings",
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"../api/transport:datagram_transport_interface",
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"../api/transport/media:media_transport_interface",
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"../api/transport/rtp:rtp_source",
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"../api/units:data_rate",
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"../api/video:video_bitrate_allocation",
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"../api/video:video_bitrate_allocator_factory",
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"../api/video:video_codec_constants",
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"../api/video:video_frame",
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"../api/video:video_frame_i420",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:rtc_software_fallback_wrappers",
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"../api/video_codecs:video_codecs_api",
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"../call",
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"../call:call_interfaces",
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"../call:video_stream_api",
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"../common_video",
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"../modules/audio_device",
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"../modules/audio_device:audio_device_impl",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/audio_processing:api",
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"../modules/audio_processing/aec_dump",
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"../modules/audio_processing/agc:gain_control_interface",
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"../modules/video_coding",
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"../modules/video_coding:video_codec_interface",
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"../modules/video_coding:video_coding_utility",
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"../rtc_base",
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"../rtc_base:audio_format_to_string",
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"../rtc_base:checks",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:stringutils",
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"../rtc_base/experiments:experimental_screenshare_settings",
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"../rtc_base/experiments:field_trial_parser",
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"../rtc_base/experiments:min_video_bitrate_experiment",
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"../rtc_base/experiments:normalize_simulcast_size_experiment",
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"../rtc_base/experiments:rate_control_settings",
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"../rtc_base/system:rtc_export",
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"../rtc_base/third_party/base64",
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"../system_wrappers",
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"../system_wrappers:field_trial",
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"../system_wrappers:metrics",
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"//third_party/abseil-cpp/absl/algorithm:container",
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"//third_party/abseil-cpp/absl/strings",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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sources = [
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"engine/adm_helpers.cc",
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"engine/adm_helpers.h",
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"engine/null_webrtc_video_engine.h",
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"engine/payload_type_mapper.cc",
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"engine/payload_type_mapper.h",
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"engine/simulcast.cc",
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"engine/simulcast.h",
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"engine/unhandled_packets_buffer.cc",
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"engine/unhandled_packets_buffer.h",
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"engine/webrtc_media_engine.cc",
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"engine/webrtc_media_engine.h",
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"engine/webrtc_video_engine.cc",
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"engine/webrtc_video_engine.h",
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"engine/webrtc_voice_engine.cc",
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"engine/webrtc_voice_engine.h",
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]
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public_configs = []
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if (!build_with_chromium) {
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public_configs += [ ":rtc_media_defines_config" ]
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deps += [ "../modules/video_capture:video_capture_internal_impl" ]
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}
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if (rtc_enable_protobuf) {
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deps += [ "../modules/audio_processing/aec_dump:aec_dump_impl" ]
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} else {
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deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ]
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}
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}
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# Heavy but optional helper for unittests and webrtc users who prefer to use
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# defaults factories or do not worry about extra dependencies and binary size.
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rtc_library("rtc_media_engine_defaults") {
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visibility = [ "*" ]
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allow_poison = [
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"audio_codecs",
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"default_task_queue",
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"software_video_codecs",
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]
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sources = [
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"engine/webrtc_media_engine_defaults.cc",
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"engine/webrtc_media_engine_defaults.h",
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]
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deps = [
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":rtc_audio_video",
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"../api/audio_codecs:builtin_audio_decoder_factory",
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"../api/audio_codecs:builtin_audio_encoder_factory",
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"../api/task_queue:default_task_queue_factory",
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"../api/video:builtin_video_bitrate_allocator_factory",
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"../api/video_codecs:builtin_video_decoder_factory",
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"../api/video_codecs:builtin_video_encoder_factory",
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"../modules/audio_processing:api",
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"../rtc_base:checks",
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]
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}
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rtc_library("rtc_data") {
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defines = [
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# "SCTP_DEBUG" # Uncomment for SCTP debugging.
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]
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deps = [
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":rtc_media_base",
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"..:webrtc_common",
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"../api:call_api",
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"../api:transport_api",
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"../p2p:rtc_p2p",
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"../rtc_base",
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"../rtc_base:rtc_base_approved",
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"../rtc_base/third_party/sigslot",
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"../system_wrappers",
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"//third_party/abseil-cpp/absl/algorithm:container",
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"//third_party/abseil-cpp/absl/base:core_headers",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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if (rtc_enable_sctp) {
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sources = [
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"sctp/sctp_transport.cc",
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"sctp/sctp_transport.h",
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"sctp/sctp_transport_internal.h",
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]
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} else {
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# libtool on mac does not like empty targets.
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sources = [
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"sctp/noop.cc",
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]
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}
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if (rtc_enable_sctp && rtc_build_usrsctp) {
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include_dirs = [
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# TODO(jiayl): move this into the public_configs of
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# //third_party/usrsctp/BUILD.gn.
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"//third_party/usrsctp/usrsctplib",
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]
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deps += [ "//third_party/usrsctp" ]
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}
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}
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rtc_source_set("rtc_media") {
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visibility = [ "*" ]
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allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
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deps = [
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":rtc_audio_video",
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":rtc_data",
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]
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}
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if (rtc_include_tests) {
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rtc_library("rtc_media_tests_utils") {
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testonly = true
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defines = []
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deps = [
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":rtc_audio_video",
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":rtc_internal_video_codecs",
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":rtc_media",
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":rtc_media_base",
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":rtc_simulcast_encoder_adapter",
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"../api:call_api",
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"../api:fec_controller_api",
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"../api:scoped_refptr",
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"../api/video:encoded_image",
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"../api/video:video_bitrate_allocation",
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"../api/video:video_frame",
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"../api/video:video_frame_i420",
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"../api/video:video_rtp_headers",
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"../api/video_codecs:video_codecs_api",
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"../call:call_interfaces",
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"../call:mock_rtp_interfaces",
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"../call:video_stream_api",
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"../common_video",
|
|
"../modules/audio_processing",
|
|
"../modules/audio_processing:api",
|
|
"../modules/rtp_rtcp:rtp_rtcp_format",
|
|
"../modules/video_coding:video_codec_interface",
|
|
"../modules/video_coding:video_coding_utility",
|
|
"../p2p:rtc_p2p",
|
|
"../rtc_base",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:gunit_helpers",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:stringutils",
|
|
"../rtc_base/third_party/sigslot",
|
|
"../test:test_support",
|
|
"//testing/gtest",
|
|
"//third_party/abseil-cpp/absl/algorithm:container",
|
|
"//third_party/abseil-cpp/absl/strings",
|
|
]
|
|
sources = [
|
|
"base/fake_frame_source.cc",
|
|
"base/fake_frame_source.h",
|
|
"base/fake_media_engine.cc",
|
|
"base/fake_media_engine.h",
|
|
"base/fake_network_interface.h",
|
|
"base/fake_rtp.cc",
|
|
"base/fake_rtp.h",
|
|
"base/fake_video_renderer.cc",
|
|
"base/fake_video_renderer.h",
|
|
"base/test_utils.cc",
|
|
"base/test_utils.h",
|
|
"engine/fake_webrtc_call.cc",
|
|
"engine/fake_webrtc_call.h",
|
|
"engine/fake_webrtc_video_engine.cc",
|
|
"engine/fake_webrtc_video_engine.h",
|
|
]
|
|
}
|
|
|
|
rtc_media_unittests_resources = [
|
|
"../resources/media/captured-320x240-2s-48.frames",
|
|
"../resources/media/faces.1280x720_P420.yuv",
|
|
"../resources/media/faces_I420.jpg",
|
|
"../resources/media/faces_I422.jpg",
|
|
"../resources/media/faces_I444.jpg",
|
|
"../resources/media/faces_I411.jpg",
|
|
"../resources/media/faces_I400.jpg",
|
|
]
|
|
|
|
if (is_ios) {
|
|
bundle_data("rtc_media_unittests_bundle_data") {
|
|
testonly = true
|
|
sources = rtc_media_unittests_resources
|
|
outputs = [
|
|
"{{bundle_resources_dir}}/{{source_file_part}}",
|
|
]
|
|
}
|
|
}
|
|
|
|
rtc_test("rtc_media_unittests") {
|
|
testonly = true
|
|
|
|
defines = []
|
|
deps = [
|
|
":rtc_audio_video",
|
|
":rtc_constants",
|
|
":rtc_data",
|
|
":rtc_encoder_simulcast_proxy",
|
|
":rtc_internal_video_codecs",
|
|
":rtc_media",
|
|
":rtc_media_base",
|
|
":rtc_media_engine_defaults",
|
|
":rtc_media_tests_utils",
|
|
":rtc_simulcast_encoder_adapter",
|
|
":rtc_vp9_profile",
|
|
"../:webrtc_common",
|
|
"../api:create_simulcast_test_fixture_api",
|
|
"../api:fake_media_transport",
|
|
"../api:libjingle_peerconnection_api",
|
|
"../api:mock_video_bitrate_allocator",
|
|
"../api:mock_video_bitrate_allocator_factory",
|
|
"../api:mock_video_codec_factory",
|
|
"../api:mock_video_encoder",
|
|
"../api:rtp_parameters",
|
|
"../api:scoped_refptr",
|
|
"../api:simulcast_test_fixture_api",
|
|
"../api/audio_codecs:builtin_audio_decoder_factory",
|
|
"../api/audio_codecs:builtin_audio_encoder_factory",
|
|
"../api/rtc_event_log",
|
|
"../api/task_queue",
|
|
"../api/task_queue:default_task_queue_factory",
|
|
"../api/test/video:function_video_factory",
|
|
"../api/transport/media:media_transport_interface",
|
|
"../api/units:time_delta",
|
|
"../api/video:builtin_video_bitrate_allocator_factory",
|
|
"../api/video:video_bitrate_allocation",
|
|
"../api/video:video_frame",
|
|
"../api/video:video_frame_i420",
|
|
"../api/video:video_rtp_headers",
|
|
"../api/video_codecs:builtin_video_decoder_factory",
|
|
"../api/video_codecs:builtin_video_encoder_factory",
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../audio",
|
|
"../call:call_interfaces",
|
|
"../common_video",
|
|
"../modules/audio_device:mock_audio_device",
|
|
"../modules/audio_processing",
|
|
"../modules/audio_processing:api",
|
|
"../modules/audio_processing:mocks",
|
|
"../modules/rtp_rtcp",
|
|
"../modules/video_coding:simulcast_test_fixture_impl",
|
|
"../modules/video_coding:video_codec_interface",
|
|
"../modules/video_coding:webrtc_vp8",
|
|
"../p2p:p2p_test_utils",
|
|
"../rtc_base",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:gunit_helpers",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../rtc_base:rtc_task_queue",
|
|
"../rtc_base:stringutils",
|
|
"../rtc_base/experiments:min_video_bitrate_experiment",
|
|
"../rtc_base/third_party/sigslot",
|
|
"../test:audio_codec_mocks",
|
|
"../test:field_trial",
|
|
"../test:rtp_test_utils",
|
|
"../test:test_main",
|
|
"../test:test_support",
|
|
"../test:video_test_common",
|
|
"//third_party/abseil-cpp/absl/algorithm:container",
|
|
"//third_party/abseil-cpp/absl/memory",
|
|
"//third_party/abseil-cpp/absl/strings",
|
|
"//third_party/abseil-cpp/absl/types:optional",
|
|
]
|
|
sources = [
|
|
"base/codec_unittest.cc",
|
|
"base/rtp_data_engine_unittest.cc",
|
|
"base/rtp_utils_unittest.cc",
|
|
"base/stream_params_unittest.cc",
|
|
"base/turn_utils_unittest.cc",
|
|
"base/video_adapter_unittest.cc",
|
|
"base/video_broadcaster_unittest.cc",
|
|
"base/video_common_unittest.cc",
|
|
"engine/encoder_simulcast_proxy_unittest.cc",
|
|
"engine/internal_decoder_factory_unittest.cc",
|
|
"engine/multiplex_codec_factory_unittest.cc",
|
|
"engine/null_webrtc_video_engine_unittest.cc",
|
|
"engine/payload_type_mapper_unittest.cc",
|
|
"engine/simulcast_encoder_adapter_unittest.cc",
|
|
"engine/simulcast_unittest.cc",
|
|
"engine/unhandled_packets_buffer_unittest.cc",
|
|
"engine/webrtc_media_engine_unittest.cc",
|
|
"engine/webrtc_video_engine_unittest.cc",
|
|
]
|
|
|
|
# TODO(kthelgason): Reenable this test on iOS.
|
|
# See bugs.webrtc.org/5569
|
|
if (!is_ios) {
|
|
sources += [ "engine/webrtc_voice_engine_unittest.cc" ]
|
|
}
|
|
|
|
if (rtc_enable_sctp) {
|
|
sources += [
|
|
"sctp/sctp_transport_reliability_unittest.cc",
|
|
"sctp/sctp_transport_unittest.cc",
|
|
]
|
|
}
|
|
|
|
if (rtc_opus_support_120ms_ptime) {
|
|
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
|
|
} else {
|
|
defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
|
|
}
|
|
|
|
data = rtc_media_unittests_resources
|
|
|
|
if (is_android) {
|
|
deps += [ "//testing/android/native_test:native_test_support" ]
|
|
shard_timeout = 900
|
|
}
|
|
|
|
if (is_ios) {
|
|
deps += [ ":rtc_media_unittests_bundle_data" ]
|
|
}
|
|
}
|
|
}
|